Over the past several years there has been, and continues to be, a tremendous amount of activity in the area of efficient encoding of speech. For an evolving digital telephone network, a most important application is the possible replacement of the 64,000 bit-per-second (b/s) PCM signal (8 bits per time slot, repeated at an 8 kHz rate) with other coding algorithms for telphoney--both in the public switched and private line networks. The reason, of course, is to achieve bandwidth compression.
For a realistic mix of input speech, music and voiceband data, adaptive differential PCM appears to be a most promising approach. One form of adaptive differential PCM coding is disclosed, for example, in U.S. Pat. No. 4,437,087 issued Mar. 13, 1984 to D. W. Petr, and can be considered a benchmark since a single encoding with this ADPCM coder at 32 kb/s is near to being subjectively equivalent to 64 kb/s .mu.255 PCM.
The prior adaptive predictors employed in the Petr coding arrangement are particularly advantageous in handling signals encountered in conventional telephone channels, i.e., signals up to 4 kHz. However, processing of so-called wideband signals, i.e., signals above 4 kHz, imposes more stringent requirements on the adaptive predictor. Indeed, wideband processing is intended to provide reduced distortion and noise with greater audio bandwidth as compared to conventional telephone transmission.
Prior predictor arrangements have performance limitations in the presence of transmission errors which result in a decoder not tracking a corresponding coder. Such mistracking results in degradation of the received signal. Moreover, wideband applications typically require so-called higher order filters in the predictor. Heretofore, maintaining tracking and stability of the higher order predictor filters was a problem.